This invention relates to Internet communications, but more specifically to a method and an apparatus to assure quality of service (“QoS”) of voice or constant bit-rate data at a guaranteed-bandwidth across multiple IP networks, including networks operated by different carriers.
Internet devices use an SIP protocol to establish a session between two users over an IP network. This may involve a multimedia session or a VoIP telephone voice call. Voice calls have a very stringent requirement for delay and jitter to preserve high quality. If the caller and called parties' IP addresses are located in the same carrier's domain, it is easy to engineer the network to meet QoS requirements. If, however, the caller and called party are using different carrier's network, it is difficult to realize whether the link for the established call spanning multiple networks will meet desired quality requirements.
IETF protocol SIP (RFC 3261) describes how to set up a voice call in an IP environment. SIP header contains information about caller, caller, proxy gateways etc. while the SIP body contains information on codec used, type of media, sampling rate used etc. using SDP (Session Description Protocol). But there is typically no framework in a conventional IP network to negotiate end-end QoS when the call is traveling two or more administrative domains (Operators). Negotiating end-to-end QoS will help carriers provide consistent quality for voice calls or multimedia sessions. In cases where consistent quality can not be maintained, the caller and caller may be notified that the call can be set-up but might suffer degraded quality.
VoIP (Voice over Internet Protocol) has a potential for saving significant long haul data transport costs. VoIP implementing SIP (session Internet protocol) is a standard mechanism for supporting voice services and multimedia sessions over IP networks. GSM and UMTS Rel. 5 standards are also based on IP standards and, as such, may conveniently be integrated with IP networks. Because many carriers in United States and elsewhere currently deploy private IP networks to bypass long distance charges of legacy interchange carriers, VoIP voice calls or multimedia sessions may likely become a preferred mode of data transport.
At present, it is commonplace to use IXCs (Interexchange Carriers) such as those provided by AT&T, Sprint and MCI, to carry long distance calls. These providers deliver calls between and among LECs/WSPs (Local Exchange Carriers/Wireless Service Providers. When the IXCs and LECs/WSPs use VoIP or multimedia sessions (e.g., streaming audio or video) in their respective networks, a mechanism to guarantee end-to-end call quality needs to be established.
As known in the art, voice calls are extremely sensitive to delay and jitter. Telecommunication standards specify that end-to-end latency for a voice call preferably should not exceed about 250 to 400 ms (milliseconds). It has been observed in laboratory environments, however, that degradation of call quality becomes noticeable when jitter or delay exceeds 350 ms. In a mobile to mobile telephone call, for example, delay on the access side of the base station is typically 200 ms. Additional delay results from coding, decoding, and the buffering. Thus, in this case, the network provider must complete the end-to-end call within the remaining 150 ms to stay within acceptable delay tolerance.
A call or multimedia session may transgress multiple independently operated private IP networks. As indicated above, call quality may conveniently be maintained between VoIP mobile telephones of a common carrier sharing the same network since the carrier has direct control over switching and routing. If, however, a call is made from a first subscriber of a first VoIP network to a second subscriber of a second independent VoIP network who may be roaming near a distant network away from his or her home location, bearer traffic must be forwarded from the first to second network and then handed over to the distant network for forwarding to the called party. In such a scenario, the total delay may likely exceed 350 ms for the mobile-to-mobile call. Delay is just one example of call degradation but other quality factors such as jitter, the effect of multiple echo cancellation, etc. may also degrade the call.
In view of the foregoing, the present invention aims to improve call quality conveyed over multiple or independently operated IP network, private or public, thereby to improve message transport quality.
The present invention also aims to improve call quality by enabling a subscriber or his/her equipment, to accept or reject a call through multiple independently operated IP networks based on a minimum acceptable delay or other factors.
It is another feature of the present invention to enable testing of transport quality that may be experienced in a session involving multiple IP networks.
It is another feature of the present invention to ensure call quality when a call travels through multiple networks provided by different service providers or network operators.
In addition to voice calls, the inventive concepts may be adopted for multimedia sessions including music or video transmissions where IP network characteristics or the transport route may impair the data transport quality.